Hi, I'm working on Asterisk and would like to use only Asterisk SIP signalling for my Voip application.
I have written my own channel driver and want to integrate my own RTP with Asterisk. SIP signalling is working fine. But i could not find API's to get RTP Port and IP address to start without starting rtp session. The only way I found to receive/send rtp information is by creating a rtp session from channel driver using ast_rtp_new_with_bindaddr( ); which means using asterisk rtp stack. This is not what i want. If anyone has done RTP integration with Asterisk earlier please help me. Thanks in advance. Garge.
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